I also looked at Naver SIP-related open-source projects, and Mobile SIP also opened SipServlet, a basic Java API, at Oracle Homepage, but I don't understand it clearly. Of course, I think it is impossible to develop it without any reason, and I need to use open source that has already been developed, but I don't know how to operate basic SIP, so I don't understand it more.ㅠ<
I think it's hard to burn yourself I'm just trying to do the audio.
Client1 After recording for 2 seconds, transfer to server via HTTP + download client 2 recording file from server and play it / After recording for 2 seconds, transfer to server via HTTP + download client 2 recording file from server and play it / After recording for 2 seconds, transfer to server via HTTP + download client 2 recording file from server and play it / ...
Client2 After recording for 2 seconds, transfer to server via HTTP + download client 1 off recording file from server and play it / After recording for 2 seconds, transfer to server via HTTP + download client 1 off recording file from server and play it / After recording for 2 seconds, transfer to server via HTTP + download client 1 off recording file from server and play it / ... Can't we just implement it like this? I feel stuffy
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I think you should understand the overall concept of SIP and understand the part.
Voice data transmission requires the use of RTP
specialized in real-time data processing.
I recommend a site that explains the concept well.
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